Historically, in the field of data communications, modems, data service units (DSU's), or channel service units (CSU's) have been used to convey information from one location to another. Digital technology now enables other communication devices, such as frame relay data service units (DSU's) and frame relay access units (FRAU's) to communicate large amounts of data at higher speeds. The communication scheme employed by these devices generally adheres to a model, known as the Open Systems Interconnect (OSI) Seven-Layer model. This model specifies the parameters and conditions under which information is formatted and transferred over a given communications network. A general background of the OSI seven-layer model follows.
In 1978, a framework of international standards for computer network architecture known as “OSI” (Open Systems Interconnect) was developed. The OSI reference model of network architecture consists of seven layers. From the lowest to the highest, the layers are: (1) the physical layer; (2) the datalink layer; (3) the network layer; (4) the transport layer; (5) the session layer; (6) the presentation layer; and (7) the application layer. Each layer uses the layer below it to provide a service to the layer above it. The lower layers are implemented by lower level protocols which define the electrical and physical standards, perform the byte ordering of the data, and govern the transmission, and error detection and correction of the bit stream. The higher layers are implemented by higher level protocols which deal with, inter alia, data formatting, terminal-to-computer dialogue, character sets, and sequencing of messages.
Layer 1, the physical layer, controls the direct host-to-host communication between the hardware of the end users' data terminal equipment (e.g., a modem connected to a PC).
Layer 2, the datalink layer, generally fragments the data to prepare it to be sent on the physical layer, receives acknowledgment frames, performs error checking, and re-transmits frames which have been incorrectly received.
Layer 3, the network layer, generally controls the routing of packets of data from the sender to the receiver via the datalink layer, and it is used by the transport layer. An example of the network layer is the Internet Protocol (IP), which is the network layer for the TCP/IP protocol widely used on Ethernet networks. In contrast to the OSI seven-layer architecture, TCP/IP (Transmission Control Protocol over Internet Protocol) is a five-layer architecture which generally consists of the network layer and the transport layer protocols.
Layer 4, the transport layer, determines how the network layer should be used to provide a point-to-point, virtual, error-free connection so that the end point devices send and receive uncorrupted messages in the correct order. This layer establishes and dissolves connections between hosts. It is used by the session layer. TCP is an example of the transport layer.
Layer 5, the session layer, uses the transport layer and is used by the presentation layer. The session layer establishes a connection between processes on different hosts. It handles the creation of sessions between hosts as well as security issues.
Layer 6, the presentation layer, attempts to minimize the noticeability of differences between hosts and performs functions such as text compression, and format and code conversion.
Layer 7, the application layer, is used by the presentation layer to provide the user with a localized representation of data which is independent of the format used on the network. The application layer is generally concerned with the user's view of the network and generally deals with resource allocation, network transparency and problem partitioning.
The communications networks that operate within the OSI seven-layer model include a number of paths or links that are interconnected to route voice, video, and/or digital data (hereinafter, collectively referred to as “data”) traffic from one location of the network to another. At each location, an interconnect node couples a plurality of source nodes and destination nodes to the network. In some cases, the sources and destinations are incorporated in a private line network that may include a series of offices connected together by leased-lines with switching facilities and transmission equipment owned and operated by the carrier or service provider and leased to the user.
This type of network is conventionally referred to as a “circuit-switching network”. Accordingly, a source node of one office at one location of the network may transmit data to a destination node of a second office located at another location of the network through their respective switching facilities.
At any given location, a large number of source nodes may desire to communicate through their respective switching facilities, or interconnect node, to destination nodes at various other locations of the network. The data traffic from the various source nodes is first multiplexed through the source switching facility, then demultiplexed at the destination switching facility, and finally delivered to the proper destination node. A variety of techniques for efficiently multiplexing data from multiple source nodes onto a single circuit of the network are presently employed in private line networks. For instance, time division multiplexing (TDM) affords each source node full access to the allotted bandwidth of the circuit for a small amount of time. The circuit is divided into defined time segments, with each segment corresponding to a specific source node, to provide for the transfer of data from those source nodes, when called upon, through the network.
Another type of network is conventionally referred to as a “packet switching network”. Frame-relay networks are one implementation of a packet-switching network. Packet-switching networks, as opposed to circuit-switching networks, allow multiple users to share data network facilities and bandwidth, rather than providing a specific amount of dedicated bandwidth to each user, as in TDM. Instead, packet switches divide bandwidth into connectionless, virtual circuits. Virtual circuits can be permanent virtual circuits (PVC's) or switched virtual circuits (SVC's). As is known, virtual circuit bandwidth is consumed only when data is actually transmitted. Otherwise, the bandwidth is not used. In this way, packet-switching networks essentially mirror the operation of a statistical multiplexer (whereby multiple logical users share a single network access circuit). Frame relay generally operates within layer 2 (the data link layer) of the OSI model, and is an improvement over previous packet switching techniques, such as the industry standard X.25, in that frame relay requires significantly less overhead.
In frame relay networks, as in all communication networks, access to the network is provided by a network service provider. These service providers generally provide the communication and switching facilities over which the above-mentioned communication devices operate. Typically, an end user desirous of establishing a communications network, provisions, or obtains from a network service provider, network services in the form of a public switched service network. An example of a public switched network is the public switched telephone network (PSTN) or a public data network (PDN). These public networks typically sell network services, in the form of connectivity, to end users.
Typically an end user of a public network will purchase a particular level of service from the network service provider. This level of service can be measured by, for example, network availability as a percentage of total time on the network, the amount of data actually delivered through the network compared to the amount of data attempted or possibly the network latency, or the amount of time it takes for a particular communication to traverse the network.
Service providers typically provide service level agreements (SLA) that guarantee certain levels of performance to end-users, including the rate of data flow, transmission delays, and availability of service. Frame relay networks typically provide for the transport of information within a “committed” information rate (CIR), as well as allowing for bursts of data above the committed rate, which is called the burst excess rate (Be). The CIR is the rate at which the network agrees to transfer information under normal conditions. The CIR is the number of bits of information transferred by the network over a specific time interval (Tc), and is typically expressed in bits per second (bps). The committed burst size (Bc) is the maximum amount of data (in bits) that a network agrees to transfer under normal conditions over measurement interval Tc. The relationship between these parameters is CIR=Bc/Tc. The excess burst size (Be) is the maximum amount of uncommitted data (in bits) that the network will attempt to deliver over measurement interval Tc. Any bits that exceed B, +Be bits in a given measurement interval (Tc) are subject to discard by the network.
Typically, data transmitted below CIR is guaranteed by the SLA. Although data transmission above CIR is available and even encouraged, it typically is not guaranteed by the SLA. Instead, the network typically makes a best effort to deliver the burst data, as long as there is excess capacity available in the network (i.e., where there are multiple PVCs on a given physical connection, if one PVC is exhibiting very low data traffic, another PVC can burst over its CIR). When capacity is not available, however, information above the guaranteed CIR is generally discarded by the network, leaving the end user responsible for re-transmitting the data. As an example, a network service provider may offer an SLA that guarantees a 99.5% data delivery ratio (DDR) (i.e., the ratio of data received to data transmitted) for data packets transmitted within the CIR but only a 75% DDR for data packets transmitted above the CIR. These guarantees are based on the fact that during periods of network congestion data packets that exceed the CIR are discarded first in order to alleviate the congestion and decrease the likelihood that the congestion will result in the discard of data packets transmitted within the CIR.
Because data transmission above the CIR may result in data being discarded, proper sizing of the CIR is a key planning parameter for both service providers and end users. One problem with current communication systems is that it is difficult for an end user to adequately determine whether the service provider is meeting its guarantee with respect to data transmission below the CIR. A related problem occurs when the end user's traffic patterns increase to a level exceeding the guaranteed CIR, which can result in data being discarded by the network due to an undersized CIR.
To the extent that current communication systems address these problems at all, such systems are limited to informing the user, in the simplest possible terms, whether or not a PVC needs a larger CIR. At best, such systems merely determine if the CIR is exceeded. If the CIR is exceeded, these systems assume that the CIR should be increased and recommend a non-specific “increase” in the CIR. However, this assumption is often inappropriate for many end user situations because the end user's objectives are not considered. Current systems fail to account for the fact that some users have a higher tolerance for lost data than other users. Furthermore, current systems, whether automatic or manual, are incapable of determining and recommending to the user the amount by which the CIR should be increased. Similarly, there are no current systems, whether automatic or manual, by which the optimum adjustments to the contracted quality of service may be determined dynamically based on the actual performance of the system. Therefore, there is a need in the industry for a system and method that automatically analyze historical data and recommend adjustments to the network service parameters based on such analysis.